[OpenSIPS-Users] Call Pickup not working for Outbound to Inbound calls

osiris123d duane.larson at gmail.com
Thu Jul 8 00:26:24 CEST 2010


You are absolutely right.

Here is the sip message I posted yesterday of User B trying to grab the call

U 2010/07/05 21:35:23.270298 173.x.x.134:5060 -> 64.2.142.93:5060 
INVITE sip:[hidden email]:5060 SIP/2.0. 
Record-Route: 
<sip:173.x.x.134;lr=on;ftag=oydxjqj42f;nat=yes;did=dc3.d5d7dc11>. 
Via: SIP/2.0/UDP 173.x.x.134;branch=z9hG4bK104b.eef8e637.0. 
Via: SIP/2.0/UDP 
192.168.0.12:2077;received=75.x.x.158;branch=z9hG4bK-whzk6bav7ueb;rport=1025. 
From: "Blah 2001" <sip:[hidden email]>;tag=oydxjqj42f. 
To: <sip:[hidden email]>. 
Call-ID: 3c27c0c40a8e-c8exk4s3gwo5. 
CSeq: 2 INVITE. 
Max-Forwards: 69. 
Contact: <sip:[hidden email]:1025;line=dkpvlkre>;reg-id=1. 
Replaces: 
[hidden email];to-tag=as55840b13;from-tag=5667c613. 


And here is the actual first invite that comes from the PSTN that I didn't
originally post

U 2010/07/05 21:35:18.642951 64.2.142.15:5060 -> 173.x.x.134:5060
INVITE sip:9xx2xx2xx09 at 173.x.x.134 SIP/2.0.
Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK19694dc7;rport.
From: "9xx8xx3xx2" <sip:9xx8xx3xx2 at 64.2.142.15>;tag=as55840b13.
To: <sip:9xx2xx2xx09 at 173.x.x.134>.
Contact: <sip:9xx8xx3xx2 at 64.2.142.15>.
Call-ID: 4df4f89829fad0997025b2d5532d7bdf at 64.2.142.15.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Date: Tue, 06 Jul 2010 02:35:19 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 332.
.
v=0.
o=root 15670 15670 IN IP4 64.2.142.15.
s=session.
c=IN IP4 64.2.142.15.
t=0 0.
m=audio 14536 RTP/AVP 0 8 3 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.



So the first initial invite comes from Vitelity's 64.2.142.15 Asterisk PBX,
but my Call Pickup Invite gets sent to Vitelity's 64.2.142.93 Asterisk PBX. 
I will have to play with this in the script logic to make sure the invite is
sent back to the original requestor.

Thanks for the clue.
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